Digital networks are currently used to transmit, and/or store where convenient, digitally encoded voice signals. For that purpose, each voice signal to be considered is, originally, sampled and each sample digitally encoded into binary bits. In theory, at least, the higher the number of bits used to code each sample the better the coding, that is the closest the voice signal would be when decoded before being provided to the end user. Unfortunately, for the network to be efficient from an economical stand point, the traffic or in other words the number of connected users acceptable without network congestion needs be maximized. This is one of the reasons why methods have been provided for lowering the voice coding bit rates while keeping the coding distortion (noise) at acceptable levels, rather than dropping users when traffic increases over a network. It looks reasonable to improve the voice coding quality when the traffic permits it and if needed lower said quality to a predetermined acceptable level under high traffic conditions. This switching from one quality (one bit rate) to another, should be made as simple and quick as possible at any node within the network. For that purpose, multirate coders should provide frames with embedded bit streams whereby switching from one predetermined bit rate to a lower predetermined rate would simply require dropping a predetermined portion of the frame.